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AgeCommit message (Collapse)Author
2008-04-28Fix deadlock issue in chan_zap with libss7 due to channel variables being ↵Matthew Fredrickson
set with the channel pvt lock being held. #12512 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-28Merged revisions 114708 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines When modules are embedded, they take on a different name, without the ".so" extension. Specifically check for this name, when we're checking if a module is loaded. (Closes issue #12534) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-27s/chan_zap/chan_skinny/Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-27Make MWI in chan_skinny event based modeled after chan_zap and chan_mgcp.Michiel van Baak
(closes issue #12214) Reported by: DEA Patches: chan_skinny-vm-events-v3.txt uploaded by DEA (license 3) Tested by: DEA and me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-26Unleak referenceTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-26Add 'sip qualify peer <peer>' command (with AMI SIPqualifypeer)Tilghman Lesher
(closes issue #12524) Reported by: ctooley Patches: sip_qualify_peer.diff.2 uploaded by ctooley (license 136) some modifications for trunk by Corydon76 Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-25Merged revisions 114673 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008) | 3 lines Use consistent logic for checking to see if a call number has been chosen yet. Also, remove some redundant logic I recently added in a fix. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-25Speaking of building...Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Pass the hangup cause all the way to the calling app/channel.Michiel van Baak
(closes issue #11328) Reported by: rain Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14) brought up-to-date to trunk by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Hey look, it builds.Joshua Colp
(closes issue #12519) Reported by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Merged revisions 114632 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines Re-invite RTP during a masquerade so that, for instance, an AMI redirect of two channels which are natively bridged will preserve audio on both channels. This prevents a problem with Asterisk not re-inviting due to one of the channels having being a zombie. (closes issue #12513) Reported by: mneuhauser Patches: asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Merged revisions 114624 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr 2008) | 10 lines Resolve a deadlock in chan_local by releasing the channel lock temporarily. (closes issue #11712) Reported by: callguy Patches: 11712.patch uploaded by putnopvut (license 60) Tested by: acunningham ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Merged revisions 114621 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008) | 4 lines Ensure that when we set the accountcode, it actually shows up in the CDR. (Fix for AMI Originate) (Closes issue #12007) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Merged revisions 51989 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #12496) Reported by: daniele Patches: misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471) Tested by: daniele Technically, I didn't use the patch above except to find out what revision to merge - but it's the same thing as this revision. ........ r51989 | crichter | 2007-01-24 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line added fix from #8899 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Merged revisions 114608 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008) | 4 lines Fix a silly mistake in a change I made yesterday that caused chan_iax2 to blow up very quickly. (issue #12515) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Merged revisions 114603 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines Only have one max-forwards header in outbound REFERs. Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Change a verbose message to debug.Russell Bryant
(closes issue #12514) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-23Merged revisions 114587 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008) | 2 lines Fix find_callno_locked() to actually return the callno locked in some more cases. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-23Merged revisions 114584 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 lines Add 502 support for both directions, not only one... (see r114571) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22Merged revisions 114571 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) | 2 lines Treat a 502 just like a 503, when it comes to processing a response code ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22Merged revisions 114558 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008) | 5 lines When we receive a full frame that is supposed to contain our call number, ensure that it has the correct one. (closes issue #10078) (AST-2008-006) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22Merged revisions 114537 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008) | 9 lines If the dial string passed to the call channel callback does not indicate an extension, then consider the extension on the channel before falling back to the default. (closes issue #12479) Reported by: darren1713 Patches: exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license 116) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22Add support for authenticating on a NOTIFY request. This is useful for ↵Joshua Colp
phones that require it when sending them a special packet to get them to do something (such as reload their configuration). (closes issue #9896) Reported by: IgorG Patches: sipnotify-113980-v14.patch uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22Hopefully, this will resolve the issues that russellb had with this ↵Steve Murphy
log_show_lock(). I gathered the code that filled the string, and put it in a different func which I cryptically call "append_lock_information()". Now, both log_show_lock(), and handle_show_locks() both call this code to do the work. Tested, seems to work fine. Also, log_show_lock was modified to use the ast_str stuff, along with checking for successful ast_str creation, and freeing the ast_str obj when finished. A break was inserted to terminate the search for the lock; we should never see it twice. An example usage in chan_sip.c was created as a comment, for instructional purposes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21(closes issue #6113)Jeff Peeler
Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21Add support for generic name transmission (#12484) on SS7 in chan_zapMatthew Fredrickson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21Merged revisions 114322 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call. (closes issue #12440) Reported by: aragon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-19SS7:Added - Generic Name / Access Transport / Redirecting Number handling. ↵Matthew Fredrickson
#12425 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-18Make sure ADSI is marked as unavailable on Unistim channels so voicemail ↵Joshua Colp
does not try to do some ADSI jazz. (closes issue #12460) Reported by: PerryB git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-18Merged revisions 114257 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr 2008) | 6 lines Clearing up error messages so they make a bit more sense. Also removing a redundant error message. Issue AST-15 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-18Merged revisions 114245 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr 2008) | 1 line Only complete the SIP channel name once for 'sip show channel <channel>' ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-17Thanks to snuff for finding these omissionsSteve Murphy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-16This is the scariest commit I've done in a long time. This is the ↵Steve Murphy
astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-16Merged revisions 114184 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-15Merged revisions 114148 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 lines Handle subscribe queues in all situations... Thanks to festr_ on irc for telling me about this bug. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-15Adding chanvar to SIPPEER from 1.4 branchOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-15Shorten the mac address pattern, since some phones use different identifiers ↵Jason Parker
(such as the i2050 softphone). (closes issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff uploaded by c (license 460) Tested by: c_hans git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-14Merged revisions 114120 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines The call_token on the pvt can occasionally be NULL, causing a crash. If it is NULL, we can skip this channel, since it can't the one we're looking for. (closes issue #9299) Reported by: vazir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-14During hangup it is possible for p->chan or p->owner to be NULL, so just ↵Joshua Colp
return what the channel is bridged to instead of what they are *really* bridged to. Thanks Matt Nicholson! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-14Merged revisions 114103 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 lines It is possible for the remote side to say they want T38 but not give any capabilities. (closes issue #12414) Reported by: MVF ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-12Make sure linkset is locked exiting ss7_start_callMatthew Fredrickson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-12Make sure we start incoming calls on SS7 with echo cancellation enabled. ↵Matthew Fredrickson
Also make sure when completing a COT we call ss7_start_call with the proper locks held. Lastly, make sure if we fail to get a channel from zt_new that we don't assume it's there. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-11Merged revisions 114083 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines Several places in the code called find_callno() (which releases the lock on the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen. Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed. (issue #12400) Reported by: ztel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10A 'b' option has been added which causes chan_local to return the actual ↵Joshua Colp
channel that is behind it when queried. This is useful for transfer scenarios as the actual channel will be transferred, not the Local channel. If you have been using Local channels as queue members and having issues when the agent did a blind transfer this option may solve the issue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10Merged revisions 114045 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines Be sure that we're not about to set bridgepvt NULL prior to dereferencing it. (closes issue #11775) Reported by: fujin ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10Fix spelling of existent in a few places.Joshua Colp
(closes issue #12409) Reported by: candlerb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10Merged revisions 114021 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines Don't add custom URI options if they don't exist OR they are empty. (closes issue #12407) Reported by: homesick Patches: uri_options-1.4.diff uploaded by homesick (license 91) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09Merged revisions 113927 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines We need to set the persistant_route [sic] parameter for the sip_pvt during the initial INVITE, no matter if we're building the route set from an INVITE request or response. (closes issue #12391) Reported by: benjaminbohlmann Tested by: benjaminbohlmann ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09Enable enough RTP bridging to allow P2P to work.Joshua Colp
(closes issue #11901) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-09Move all messages wrapped in skinnydebug from debug to verbose.Jason Parker
(closes issue #12224) Reported by: DEA Patches: chan_skinny-debug-log.txt uploaded by DEA (license 3) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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